Refine code

trunk/conf/full.conf

@@ -226,64 +226,134 @@ http_server {
 #############################################################################################
 # the streamer cast stream from other protocol to SRS over RTMP.
 # @see https://github.com/ossrs/srs/tree/develop#stream-architecture
+
+# MPEGTS over UDP
 stream_caster {
     # whether stream caster is enabled.
     # default: off
-    enabled         off;
+    enabled         on;
     # the caster type of stream, the casters:
     #       mpegts_over_udp, MPEG-TS over UDP caster.
-    #       rtsp, Real Time Streaming Protocol (RTSP).
-    #       flv, FLV over HTTP by POST.
     caster          mpegts_over_udp;
     # the output rtmp url.
     # for mpegts_over_udp caster, the typically output url:
     #           rtmp://127.0.0.1/live/livestream
+    output          rtmp://127.0.0.1/live/livestream;
+    # the listen port for stream caster.
+    #       for mpegts_over_udp caster, listen at udp port. for example, 8935.
+    listen          8935;
+}
+
+# RTSP
+stream_caster {
+    # whether stream caster is enabled.
+    # default: off
+    enabled         on;
+    # the caster type of stream, the casters:
+    #       rtsp, Real Time Streaming Protocol (RTSP).
+    caster          rtsp;
+    # the output rtmp url.
     # for rtsp caster, the typically output url:
     #           rtmp://127.0.0.1/[app]/[stream]
     #       for example, the rtsp url:
     #           rtsp://192.168.1.173:8544/live/livestream.sdp
     #       where the [app] is "live" and [stream] is "livestream", output is:
     #           rtmp://127.0.0.1/live/livestream
-    # for flv caster, the typically output url:
+    output          rtmp://127.0.0.1/[app]/[stream];
-    #           rtmp://127.0.0.1/[app]/[stream]
-    #       for example, POST to url:
-    #           http://127.0.0.1:8936/live/livestream.flv
-    #       where the [app] is "live" and [stream] is "livestream", output is:
-    #           rtmp://127.0.0.1/live/livestream
-    output          rtmp://127.0.0.1/live/livestream;
     # the listen port for stream caster.
-    #       for mpegts_over_udp caster, listen at udp port. for example, 8935.
     #       for rtsp caster, listen at tcp port. for example, 554.
-    #       for flv caster, listen at tcp port. for example, 8936.
+    listen          554;
-    # TODO: support listen at <[ip:]port>
-    listen          8935;
     # for the rtsp caster, the rtp server local port over udp,
     # which reply the rtsp setup request message, the port will be used:
     #       [rtp_port_min, rtp_port_max)
     rtp_port_min    57200;
     rtp_port_max    57300;
 }
+
+# FLV
 stream_caster {
-    enabled         off;
+    # whether stream caster is enabled.
-    caster          mpegts_over_udp;
+    # default: off
-    output          rtmp://127.0.0.1/live/livestream;
+    enabled         on;
-    listen          8935;
+    # the caster type of stream, the casters:
-}
+    #       flv, FLV over HTTP by POST.
-stream_caster {
-    enabled         off;
-    caster          rtsp;
-    output          rtmp://127.0.0.1/[app]/[stream];
-    listen          554;
-    rtp_port_min    57200;
-    rtp_port_max    57300;
-}
-stream_caster {
-    enabled         off;
     caster          flv;
+    # the output rtmp url.
+    # for flv caster, the typically output url:
+    #           rtmp://127.0.0.1/[app]/[stream]
+    #       for example, POST to url:
+    #           http://127.0.0.1:8936/live/livestream.flv
+    #       where the [app] is "live" and [stream] is "livestream", output is:
+    #           rtmp://127.0.0.1/live/livestream
     output          rtmp://127.0.0.1/[app]/[stream];
+    # the listen port for stream caster.
+    #       for flv caster, listen at tcp port. for example, 8936.
     listen          8936;
 }
 
+# GB28181
+stream_caster {
+    # whether stream caster is enabled.
+    # default: off
+    enabled         on;
+    # the caster type of stream, the casters:
+    #       gb28181, Push GB28181 to SRS.
+    caster          gb28181;
+    # the output rtmp url.
+    # for gb28181 caster, the typically output url:
+    #           rtmp://127.0.0.1/live/[stream]
+    #       where the [stream] is the VideoChannelCodecID.
+    output          rtmp://127.0.0.1/live/[stream];
+    # the listen port for stream caster.
+    #       for gb28181 caster, listen at udp port. for example, 9000.
+    # @remark We can bundle all gb28181 to this port, to reuse this port.
+    #       User can choose to bundle port in API port_mode or SIP invite_port_fixed.
+    listen          9000;
+    # If not bundle ports, use specified ports for each stream.
+    rtp_port_min        58200;
+    rtp_port_max        58300;
+    # Whether wait for keyframe then forward to RTMP.
+    wait_keyframe       off;
+    # Max timeout in seconds for RTP stream, if timeout, RTCP bye and close stream.
+    # default: 30
+    rtp_idle_timeout    30;
+    # Whether has audio.
+    # @remark Flash/RTMP only supports 11025 22050 44100 sample rate, if not the audio may corrupt.
+    # default: off
+    audio_enable        off;
+    # The exposed IP to receive media stream.
+    host                192.168.1.3;
+
+    sip {
+        # Whether enable embeded SIP server.
+        # default: on
+        enabled         on;
+        # The SIP listen port.
+        # default: 5060
+        listen          5060;
+        # The SIP server ID.
+        # default: 34020000002000000001
+        serial          34020000002000000001;
+        # The SIP server domain.
+        # default: 3402000000
+        realm           3402000000;
+        # The SIP ACK response timeout in seconds.
+        # default: 30
+        ack_timeout         30;
+        # The keepalive timeout in seconds.
+        # default: 120
+        keepalive_timeout   120;
+        # Whether print SIP logs.
+        print_sip_message   off;
+        # Whether play immediately after registered.
+        # default: on
+        auto_play           on;
+        # Whether bundle media stream port.
+        # default: on
+        invite_port_fixed   on;
+    }
+}
+
 #############################################################################################
 # SRT server section
 #############################################################################################

trunk/conf/push.gb28181.conf

@@ -3,8 +3,6 @@
 listen                  1935;
 max_connections         1000;
 daemon                  off;
-pid                     ./objs/srs28181.pid;
-srs_log_file            ./objs/srs28181.log;
 srs_log_tank            console;
 
 http_api {
@@ -17,6 +15,8 @@ stream_caster {
     caster              gb28181;
 
     # 转发流到rtmp服务器地址与端口
+    # TODO: https://github.com/ossrs/srs/pull/1679/files#r400875104
+    # [stream] is VideoChannelCodecID(视频通道编码ID)
     output              127.0.0.1:1935;
     
     # 接收设备端rtp流的多路复用端口
@@ -46,12 +46,13 @@ stream_caster {
     # 这样的话播放器为自动通过adts头自动选择采样频率
     # 像ffplay, vlc都可以，但是flash是没有声音，
     # 因为flash,只支持11025 22050 44100
-    audio_enable        on;
+    audio_enable        off;
 
     # 服务器主机号，可以域名或ip地址
     # 也就是设备端将媒体发送的地址，如果是服务器是内外网
     # 需要写外网地址，
     # 调用api创建stream session时返回ip地址也是host
+    # TODO: https://github.com/ossrs/srs/pull/1679/files#r400917594
     host                192.168.1.27;
 
     sip {
@@ -60,13 +61,13 @@ stream_caster {
         enabled on;
         
         # sip监听udp端口
-        listen              15060;
+        listen              5060;
         
-        #服务器端编号
+        # SIP server ID(SIP服务器ID).
         # 设备端配置编号需要与该值一致，否则无法注册
         serial              34020000002000000001;
 
-        #服务器端域
+        # SIP server domain(SIP服务器域)
         realm               3402000000;
 
         # 服务端发送ack后，接收回应的超时时间，单位为秒
@@ -75,11 +76,12 @@ stream_caster {
 
         # 设备心跳维持时间，如果指定时间内(秒）没有接收一个心跳
         # 认为设备离线
-        keepalive_timeout   30;
+        keepalive_timeout   120;
 
         # 日志打印是否打印sip信息
         # off:不打印
         # on:打印接收或发送sip命令信息
+        # TODO: https://github.com/ossrs/srs/pull/1679/files#r400929300
         print_sip_message   off;
 
         # 注册之后是否自动给设备端发送invite

trunk/src/app/srs_app_config.cpp

@@ -4310,9 +4310,9 @@ int SrsConfig::get_stream_caster_rtp_port_max(SrsConfDirective* conf)
     return ::atoi(conf->arg0().c_str());
 }
 
-int SrsConfig::get_stream_caster_gb28181_rtp_idle_timeout(SrsConfDirective* conf)
+srs_utime_t SrsConfig::get_stream_caster_gb28181_rtp_idle_timeout(SrsConfDirective* conf)
 {
-    static int DEFAULT = 30;
+    static srs_utime_t DEFAULT = 30 * SRS_UTIME_SECONDS;
     
     if (!conf) {
         return DEFAULT;
@@ -4323,7 +4323,7 @@ int SrsConfig::get_stream_caster_gb28181_rtp_idle_timeout(SrsConfDirective* conf
         return DEFAULT;
     }
     
-    return ::atoi(conf->arg0().c_str());
+    return (srs_utime_t)(::atoi(conf->arg0().c_str()) * SRS_UTIME_SECONDS);
 }
 
 int SrsConfig::get_stream_caster_gb28181_ack_timeout(SrsConfDirective* conf)
@@ -4349,7 +4349,7 @@ int SrsConfig::get_stream_caster_gb28181_ack_timeout(SrsConfDirective* conf)
 
 int SrsConfig::get_stream_caster_gb28181_keepalive_timeout(SrsConfDirective* conf)
 {
-    static int DEFAULT = 30;
+    static int DEFAULT = 120;
     
     if (!conf) {
         return DEFAULT;
@@ -4386,7 +4386,7 @@ string SrsConfig::get_stream_caster_gb28181_host(SrsConfDirective* conf)
 
 string SrsConfig::get_stream_caster_gb28181_serial(SrsConfDirective* conf)
 {
-    static string DEFAULT = "";
+    static string DEFAULT = "34020000002000000001";
     
     if (!conf) {
         return DEFAULT;
@@ -4407,7 +4407,7 @@ string SrsConfig::get_stream_caster_gb28181_serial(SrsConfDirective* conf)
 
 string SrsConfig::get_stream_caster_gb28181_realm(SrsConfDirective* conf)
 {
-    static string DEFAULT = "";
+    static string DEFAULT = "3402000000";
     
     if (!conf) {
         return DEFAULT;
@@ -4428,7 +4428,7 @@ string SrsConfig::get_stream_caster_gb28181_realm(SrsConfDirective* conf)
 
 bool SrsConfig::get_stream_caster_gb28181_audio_enable(SrsConfDirective* conf)
 {
-    static bool DEFAULT = true;
+    static bool DEFAULT = false;
     
     if (!conf) {
         return DEFAULT;
@@ -4481,7 +4481,7 @@ bool SrsConfig::get_stream_caster_gb28181_wait_keyframe(SrsConfDirective* conf)
 
 bool SrsConfig::get_stream_caster_gb28181_sip_enable(SrsConfDirective* conf)
 {
-    static bool DEFAULT = false;
+    static bool DEFAULT = true;
     
     if (!conf) {
         return DEFAULT;
@@ -4503,7 +4503,7 @@ bool SrsConfig::get_stream_caster_gb28181_sip_enable(SrsConfDirective* conf)
 
 bool SrsConfig::get_stream_caster_gb28181_sip_auto_play(SrsConfDirective* conf)
 {
-    static bool DEFAULT = false;
+    static bool DEFAULT = true;
     
     if (!conf) {
         return DEFAULT;
@@ -4525,7 +4525,7 @@ bool SrsConfig::get_stream_caster_gb28181_sip_auto_play(SrsConfDirective* conf)
 
 int SrsConfig::get_stream_caster_gb28181_sip_listen(SrsConfDirective* conf)
 {
-    static int DEFAULT = 0;
+    static int DEFAULT = 5060;
     
     if (!conf) {
         return DEFAULT;

trunk/src/app/srs_app_config.hpp

@@ -500,7 +500,7 @@ public:
     // Get the max udp port for rtp of stream caster rtsp.
     virtual int get_stream_caster_rtp_port_max(SrsConfDirective* conf);
     
-    virtual int get_stream_caster_gb28181_rtp_idle_timeout(SrsConfDirective* conf);
+    virtual srs_utime_t get_stream_caster_gb28181_rtp_idle_timeout(SrsConfDirective* conf);
     virtual int get_stream_caster_gb28181_ack_timeout(SrsConfDirective* conf);
     virtual int get_stream_caster_gb28181_keepalive_timeout(SrsConfDirective* conf);
     virtual bool get_stream_caster_gb28181_audio_enable(SrsConfDirective* conf);
@@ -514,7 +514,6 @@ public:
     virtual int get_stream_caster_gb28181_sip_listen(SrsConfDirective* conf);
     virtual bool get_stream_caster_gb28181_sip_invite_port_fixed(SrsConfDirective* conf);
 
-
 // vhost specified section
 public:
     // Get the vhost directive by vhost name.

trunk/src/app/srs_app_gb28181.cpp

@@ -1,7 +1,7 @@
 /**
  * The MIT License (MIT)
  *
- * Copyright (c) 2013-2020 Winlin
+ * Copyright (c) 2013-2020 Lixin
  *
  * Permission is hereby granted, free of charge, to any person obtaining a copy of
  * this software and associated documentation files (the "Software"), to deal in
@@ -749,7 +749,7 @@ srs_error_t SrsGb28181RtmpMuxer::do_cycle()
         }
         
         srs_utime_t now = srs_get_system_time();
-        srs_utime_t duration = (now - recv_stream_time) / (1000*1000);
+        srs_utime_t duration = now - recv_stream_time;
         
         SrsGb28181Config config = gb28181_manger->get_gb28181_config();
         if (duration > config.rtp_idle_timeout){

trunk/src/app/srs_app_gb28181.hpp

@@ -1,7 +1,7 @@
 /**
  * The MIT License (MIT)
  *
- * Copyright (c) 2013-2020 Winlin
+ * Copyright (c) 2013-2020 Lixin
  *
  * Permission is hereby granted, free of charge, to any person obtaining a copy of
  * this software and associated documentation files (the "Software"), to deal in
@@ -296,7 +296,7 @@ class SrsGb28181Config
 {
 public:
     std::string host;
-    int rtp_idle_timeout;
+    srs_utime_t rtp_idle_timeout;
     bool audio_enable;
     bool wait_keyframe;
     std::string output;

trunk/src/app/srs_app_gb28181_sip.cpp

@@ -1,7 +1,7 @@
 /**
  * The MIT License (MIT)
  *
- * Copyright (c) 2013-2020 Winlin
+ * Copyright (c) 2013-2020 Lixin
  *
  * Permission is hereby granted, free of charge, to any person obtaining a copy of
  * this software and associated documentation files (the "Software"), to deal in

trunk/src/app/srs_app_gb28181_sip.hpp

@@ -1,7 +1,7 @@
 /**
  * The MIT License (MIT)
  *
- * Copyright (c) 2013-2020 Winlin
+ * Copyright (c) 2013-2020 Lixin
  *
  * Permission is hereby granted, free of charge, to any person obtaining a copy of
  * this software and associated documentation files (the "Software"), to deal in

trunk/src/protocol/srs_sip_stack.cpp

@@ -1,7 +1,7 @@
 /**
  * The MIT License (MIT)
  *
- * Copyright (c) 2013-2020 Winlin
+ * Copyright (c) 2013-2020 Lixin
  *
  * Permission is hereby granted, free of charge, to any person obtaining a copy of
  * this software and associated documentation files (the "Software"), to deal in

trunk/src/protocol/srs_sip_stack.hpp

@@ -1,7 +1,7 @@
 /**
  * The MIT License (MIT)
  *
- * Copyright (c) 2013-2020 Winlin
+ * Copyright (c) 2013-2020 Lixin
  *
  * Permission is hereby granted, free of charge, to any person obtaining a copy of
  * this software and associated documentation files (the "Software"), to deal in